1. Field of the Invention
The present invention relates to call conferencing.
2. Description of the Related Art
Voice call conferencing is a concept well known in the art, and exists both in fixed and cellular telephony. Typically, it allows more than two parties, or communication terminals, to be involved in the same communications session. For example, when two parties are involved in a communications session, it may be desirable to invite one or more parties to the same session. This may be achieved by dialling a special code number associated with a call conferencing service activation feature, followed by the identification number of the party to be invited. A telephone switch then connects that party to the ongoing voice call between the first and the second party.
With the fast evolution of telephone networks, various communications protocols have defined new and more flexible manners of handling voice and data call sessions in telecommunications networks. The Session Initiation Protocol (SIP) is an Internet Engineering Task Force (IETF) standard protocol for initiating an interactive user session that may involve multimedia elements such as video, voice, chat, gaming, and virtual reality. Like the Hyper Text Terminal Protocol (HTTP), or the Simple Mail Transfer Protocol (SMTP), SIP works in the Application layer of the Open Systems Interconnection (OSI) communications model. SIP can establish multimedia sessions or Internet telephony calls, and modify, or terminate them. Because SIP supports name mapping and redirection services, it makes it possible for users to initiate and receive communications and services from any location, and for networks to identify the users wherever they are. SIP is a request-response protocol, dealing with requests from clients and responses from servers. Participants are usually identified by SIP Uniform Resource Locators (URLs) or Uniform Resource Identifiers (URIs), although SIP also supports E.164 telephone number addressing. Requests can be sent through any transport protocol, such as the User Datagram Protocol (UDP), the Simple Control Transport Protocol (SCTP), or the Transfer Control Protocol (TCP). SIP determines the end system to be used for the session, the communication media and media parameters, and the called party's desire to engage in the communication. Once these are assured, SIP establishes call parameters at either end of the communication, and handles call transfer and termination. The Session Initiation Protocol is specified in IETF Request for Comments (RFC) 2543, which is herein included by reference.
Regarding conference call, the Third Generation Partnership Project (3GPP) Technical Specifications (TS) 23.218 and 23.228 Release 5 describe a network reference model and a set of procedures defining how call conferencing is to be performed. Reference is now made to FIG. 1 (Prior Art), which is a high-level network reference model of a 3GPP network 100 for IP Multimedia session handling. Shown in the centre of FIG. 1 is a Serving Call State Control Function (S-CSCF) 102 that is responsible for handling the SIP call sessions and conference calls for the subscribers of network 100 as it is known in the art. S-CSCF 102 is linked via an IP Multimedia Service Control (ISC) interface 104 to an SIP Application Server (AS) 106 that executes various services for influencing and impacting the SIP sessions. Also connected to S-CSCF 102 via a similar ISC interface 104 is an IP Multimedia Service Switching Function (IM-SSF) 108, which hosts Customized Applications for Mobile network Enhanced Logic (CAMEL) network features (i.e. trigger detection points, CAMEL Service Switching Finite State Machine, etc). A CAMEL Application Part (CAP) interface 110 connects the IM-SSF 108 with a Camel Service Environment 112 as specified in 3GPP TS 29.078, herein included by reference. An Open Service Access (OSA) service capability server (SCS) 114 interfaces on one side with the S-CSCF 102 via an ISC interface 104, and to the other side to an OSA framework Application Server 116 and provides a standardized way for third party secure access to the IM subsystem. The Application Servers 106 and 108 can also interact with a Multimedia Resource Function Controller (MRFC) 118 via the S-CSCF 102 (ISC 104 and Mr 119 interfaces) in order to control Multimedia Resource Function processing. In particular, an MRFC is responsible for controlling and setting up the resources for a conference call. Finally, a Home Subscriber Server (HSS) 120 connects to the S-CSCF 102 via a Cx interface 122 and is responsible for storing and providing subscriber related information. The functioning of the network 100 for setting up and caring on SIP-based data calls and conference calls is fully described in the above-mentioned 3GPP technical specifications 23.218 and 23.228 Release 5, which are herein enclosed by reference.
However, the teaching of the above-mentioned 3GPP specifications is limited to call cases scenario wherein call conferencing is conducted between User Equipment (UE) that are within the same serving network, or between UE that are served by different serving networks but whose conference call is handled by their home network MRFC. With reference being now further made to FIG. 1, for example, when UE1 124, UE2 126, and UE3 128 are involved in a conference call, and they connect via an access network 130 to the serving network 100, the resources for the conference call are handled by the home network's MRFC 118, even in cases when one or more of the UEs are roaming outside their home network 100. This results in the multimedia traffic to be sometimes routed following lengthy routes, which results in an increase of the conferencing costs, especially in cases wherein the conferencing involves the exchange of considerable amount of data information such as for example in the case of multimedia call conferencing. The existing 3GPP technical specifications fail to teach any optimized call conferencing scenario wherein the selection of the MRFC handling conference calls may be other than the home network's MRFC.
Accordingly, it should be readily appreciated that in order to overcome the deficiencies and shortcomings of the existing solutions, it would be advantageous to have a method and system for efficiently setting up and caring on economic call session conferencing. The present invention provides such a method and system.